[asterisk-users] Problems during calls

[Sammy Govind]

Hi,

Call getting silenced in the middle definitely point to RTP but I think
the redialling part should be considered as well. I think that Phones are
loosing registrations or like Zeeshan mentioned could be getting blocked by
firewall - Asterisk server's firewall as well as any other firewall in front
of server should be inspected for sessions/connections limit etc.

--
Regards,
Sammy

On Wed, Oct 19, 2011 at 12:27 AM, Aksel Celasun <aksel at abacus-it.no> wrote:

> Thank you for the reply.****
>
> ** **
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> ** **
>
> The Asterisk is behind a firewall, but not in a dmz, been thinking of
> placing it in a dmz soon, maybe that will solve the problem.****
>
> Or else, I will try your guide with wireshark.****
>
> ** **
>
> Thank you very much.****
>
> ** **
>
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> Best regards****
>
> ** **
>
> Aksel****
>
> ** **
>
> *Fra:* asterisk-users-bounces at lists.digium.com [mailto:
> asterisk-users-bounces at lists.digium.com] *På vegne av* VisionVoIP
> *Sendt:* 18. oktober 2011 16:31
>
> *Til:* Asterisk Users Mailing List - Non-Commercial Discussion
> *Emne:* Re: [asterisk-users] Problems during calls****
>
> ** **
>
> I can only make another guess. If your system is behind a firewall, try
> adding 'insecure=invite' in your sip.conf's general section.
>
>
> To troubleshoot such cases, do a tcpdump trace like this:
>
> 1. Run tcpdump on your server before making a call. Use command "tcpdump
> port 5060 -s0 -w dumpfile.pcap".
> 2. When you notice the silence problem, hangup, and stop the trace using
> CTRL+C.
> 3. Copy the dumpfile.pcap to a computer with Wireshark installed.
> 4. Open this file in Wireshark and follow my blog at
> http://ilovetovoip.com/2010/02/graphical-illustration-of-the-sip-messages-using-wonderful-wireshark/
> 5. Given that you know some basics of how VoIP works over SIP, the
> wireshark graph will tell you if RTP was still flowing when it was silent.
> It probably is, but to which IP address.
>
> My guess is your RTP, i.e. voice date, starts flowing towards some wrong IP
> address, or stop flowing, or is blocked by the router.
>
> A good solution is to put your Asterisk server in DMZ mode.
>
> There can be many other guesses, but the above is a good start.
> --
>
> Zeeshan A Zakaria
>
> PBX - visionvoip.com
> Blog - ilovetovoip.com
>
> On 18/10/2011 10:02, Aksel Celasun wrote: ****
>
> Thank you for replying****
>
>  ****
>
>  ****
>
> My sip.conf is set to no on canreinvite****
>
>  ****
>
>  ****
>
>  ****
>
> ** **
>
> --
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